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JP2007060648

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DESCRIPTION JP2007060648
An object of the present invention is to perform more stable distortion removal processing by
performing signal processing following changes in parameters in an actual speaker. A speaker
apparatus according to the present invention performs feedforward processing on an electric
signal to be input to a speaker based on the speaker and a preset filter coefficient so as to
remove non-linear distortion generated from the speaker. A feedforward processing unit, and a
feedback processing unit that detects vibration of the speaker and performs feedback processing
on the electric signal relating to the vibration with respect to the electric signal to be input to the
speaker. The feedback processing unit generates the signal from the speaker The electric signal
related to the vibration is feedback-processed so as to remove the non-linear distortion and so
that the frequency characteristic related to the vibration of the speaker has a predetermined
frequency characteristic. [Selected figure] Figure 1
Speaker device
[0001]
The present invention relates to a speaker device, and more particularly to a speaker device that
removes distortion generated from a speaker.
[0002]
Conventionally, it is desirable to faithfully convert an electrical signal into a sound wave in a
normal speaker that does not undergo electrical signal processing.
08-05-2019
1
However, in an actual speaker, it is difficult to perform faithful conversion because of the
structural limitation. For example, in the magnetic circuit constituting the speaker, the magnetic
flux density in the magnetic gap decreases as the amplitude increases due to the structure. The
force coefficient also decreases as the magnetic flux density decreases. In addition, the stiffness
of the support system such as the damper or the edge changes in accordance with the amplitude
due to the structure of the support system. For these reasons, etc., the amplitude of the speaker is
not necessarily proportional to the magnitude of the input electrical signal, and there is a
problem that non-linear distortion occurs.
[0003]
Therefore, as a method of removing the non-linear distortion, a method using electrical signal
processing such as feed forward processing has been conventionally proposed. This processing
method is a method of polynomial-approximating a parameter including a nonlinear component
of a speaker (a force coefficient related to a magnetic flux density, stiffness of a support system,
etc.) and setting a filter coefficient so as to cancel nonlinear distortion caused by the parameter.
is there. Non-linear distortion is removed by inputting the electrical signal to the speaker through
the filter with the filter coefficient set. However, among the above-mentioned parameters, the
stiffness of the support system, in particular, varies from moment to moment depending on the
magnitude of the electrical signal input to the speaker, and also varies with age. That is, the value
of the parameter changes with time. Therefore, in the feedforward process, the error between the
preset parameter value and the actual parameter value increases with time, and the distortion
removal effect is significantly impaired.
[0004]
Therefore, in order to solve the above problems, a method has been proposed in which the
parameters of the filter coefficients are adaptively updated in feedforward processing (see, for
example, Patent Document 1). Hereinafter, this method will be described with reference to FIG.
FIG. 28 is a block diagram showing a conventional speaker device 9 that adaptively updates filter
coefficient parameters.
[0005]
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2
In FIG. 28, the conventional speaker device 9 includes a control unit 91, a parameter detector 92,
and a speaker 95. The parameter detector 92 also includes an error circuit 93 and an update
circuit 94. The error circuit 93 has a filter (not shown), and calculates pseudo vibration
characteristics from the signal input from the control unit 91 in the filter. Then, the error circuit
93 predicts and calculates the drive voltage applied to the speaker 95 from the pseudo vibration
characteristic. The predicted drive voltage is equivalent to the impedance characteristic when the
speaker 95 is current-driven. Next, the error circuit 93 generates an error signal e (t) by
subtracting the drive voltage applied to the actual speaker 95 from the predicted drive voltage.
The error signal e (t) is input to the updating circuit 94.
[0006]
The update circuit 94 calculates a parameter in the control unit 91 to be updated based on the
error signal e (t). The parameter calculated in the updating circuit 94 is reflected on the abovementioned filter in the error circuit 93, and the error circuit 93 generates the gradient signal Sg.
The gradient signal Sg generated in the error circuit 93 is output to the update circuit 94 again.
Thus, the updating circuit 94 uses the error signal e (t) and the gradient signal Sg to calculate a
parameter that minimizes the error signal e (t). The parameters at the time when the error signal
e (t) becomes minimum are output to the control unit 91 as a power vector P, and the parameters
in the control unit 91 are updated. As described above, in the speaker device 9 shown in FIG. 28,
the parameters in the error circuit 93 and the updating circuit 94 are updated such that the
parameters in the control unit 91 conform to the parameters of the actual speaker 95. Japanese
Patent Application Laid-Open No. 11-46393
[0007]
However, in the error circuit 93 and the update circuit 94 for updating the parameters described
above, complicated and extensive operations are required. In addition, as described above, the
stiffness of the support system changes from moment to moment depending on the magnitude of
the electrical signal input to the speaker. That is, in the conventional speaker device 9, it is
extremely difficult in practice to perform the updating process of the parameter in accordance
with the drastic change of the stiffness of the support system since complicated and huge
operation is required. As a result, in the conventional speaker device 9, there is a problem that
the distortion removal effect can not be sufficiently obtained and the realization is lacking. In
addition, the conventional speaker device 9 has a problem that it lacks cost performance in order
to realize a huge amount of calculation processing.
08-05-2019
3
[0008]
Therefore, an object of the present invention is to provide a speaker device capable of
performing signal processing that follows changes in parameters in an actual speaker and
performing more stable distortion removal processing.
[0009]
A first aspect of the present invention is a speaker device, which feeds-forwards an electrical
signal to be input to the speaker based on the speaker and a preset filter coefficient so as to
remove non-linear distortion generated from the speaker. A feedforward processing unit, and a
feedback processing unit that detects vibration of the speaker and performs feedback processing
on the electric signal relating to the vibration with respect to the electric signal to be input to the
speaker. The feedback processing unit generates the signal from the speaker The electric signal
related to the vibration is feedback-processed so as to remove the non-linear distortion and so
that the frequency characteristic related to the vibration of the speaker has a predetermined
frequency characteristic.
[0010]
In a second invention according to the first invention, the feedback processing unit uses an
electrical signal to be input to the speaker as an input, and converts the frequency characteristic
of the electrical signal into a predetermined frequency characteristic, and The difference between
the sensor for detecting the vibration of the speaker, the electric signal indicating the
predetermined frequency characteristic converted by the predetermined characteristic
conversion filter, and the electric signal related to the vibration detected by the sensor is
calculated, and the difference electric signal is used as an error signal. It has the 1st adder to
output, and the 2nd adder which adds the electric signal and error signal which were processed
in the feedforward processing part, and it outputs to a speaker.
[0011]
In a third invention according to the second invention, the filter coefficient in the feedforward
processing unit is a coefficient based on an inherent parameter of the speaker, and the
feedforward processing unit inputs the signal to the speaker so as to cancel the non-linear
component of the parameter. Processing the electrical signal to be done.
[0012]
In a fourth aspect based on the second aspect, the filter coefficient in the feedforward processing
unit is a factor based on a parameter unique to the speaker, and the parameter is a parameter
08-05-2019
4
that changes in accordance with the vibration displacement of the speaker. It features.
[0013]
In a fifth aspect based on the fourth aspect, the feedforward processing unit receives the
electrical signal to be input to the speaker as an input, and removes non-linear distortion
generated from the speaker based on a preset filter coefficient. And a linear filter that receives
the electric signal to be input to the speaker and generates an electric signal indicating vibration
displacement when the speaker vibrates linearly. The removal filter is characterized by referring
to an electrical signal indicative of the vibration displacement generated in the linear filter.
[0014]
According to a sixth aspect, in the fifth aspect, the filter further includes an amplifying unit
provided between the second adder and the speaker for amplifying the gain of the electrical
signal to be input to the speaker. The coefficient, the filter coefficient in the predetermined
characteristic conversion filter, and the filter coefficient in the linear filter are filter coefficients
multiplied by the reciprocal of the gain amplified in the amplification unit.
[0015]
In a seventh invention according to the fourth invention, the electric signal detected by the
sensor is an electric signal indicating vibration displacement of the speaker, and the feedforward
processing unit is an electric signal indicating vibration displacement detected in the sensor To
refer to.
[0016]
In an eighth aspect based on the second aspect, the electric signal to be input to the speaker is
provided at the front stage of the feedforward processing unit, and a predetermined frequency
characteristic is divided by a characteristic related to vibration of the speaker. It further
comprises a pre-filter that processes based on the determined filter coefficients.
[0017]
A ninth aspect of the invention according to the second aspect of the invention further comprises
limiting means for limiting the level of the electrical signal so that the electrical signal above the
predetermined level is not input to the speaker.
[0018]
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5
In a tenth aspect based on the second aspect, the feed-forward processing portion further
includes an amplification unit provided between the second adder and the speaker for amplifying
the gain of the electrical signal to be input to the speaker. The filter coefficient in the and the
filter coefficient in the predetermined characteristic conversion filter are filter coefficients
multiplied by the reciprocal of the gain amplified in the amplification unit.
[0019]
An eleventh invention is characterized in that in the first invention, the feedforward processing
unit is provided in the front stage of the speaker and is provided in a feedback loop formed by
the feedback processing unit.
[0020]
In a twelfth invention according to the first invention, the feedback processing unit uses an
electrical signal to be input to the speaker as an input, and converts the frequency characteristic
of the electrical signal into a predetermined frequency characteristic, and The difference between
the sensor for detecting the vibration of the speaker, the electric signal indicating the
predetermined frequency characteristic converted by the predetermined characteristic
conversion filter, and the electric signal related to the vibration detected by the sensor is
calculated, and the difference electric signal is used as an error signal. The feedforward
processing unit includes: a first adder to output; and a second adder to add an input electrical
signal and an error signal and output the result to a feedforward processing unit, the feedforward
processing unit including: Feedforward processing of the electric signal output from the second
adder to remove non-linear distortion generated from the speaker And outputs it to the speaker
Te.
[0021]
In a thirteenth aspect of the present invention based on the twelfth aspect, the frequency
converter is provided between the second adder and the feedforward processing unit, and the
gain of the electrical signal to be input to the speaker is in a frequency band lower than the first
frequency. The signal processing apparatus further comprises a first filter having a filter
coefficient exhibiting a characteristic inclined at -6 dB / oct, and the first frequency is a
frequency higher than the gain crossover frequency indicated by the open loop transfer
characteristic of the feedback loop formed by the feedback processing unit. It is characterized by
being.
[0022]
In a fourteenth aspect based on the twelfth aspect, the gain of the electrical signal to be input to
08-05-2019
6
the speaker is provided in a stage preceding the feedforward processing unit, and has a slope of
6 dB / oct or more in a frequency band lower than the second frequency. The signal processing
apparatus further includes a second filter having a filter coefficient indicating a sloping
characteristic, and the second frequency is a frequency higher than a gain crossover frequency
indicated by the open loop transfer characteristic of the feedback loop formed by the feedback
processing unit. I assume.
[0023]
According to a fifteenth invention, in the twelfth invention, the gain of the electric signal to be
input to the speaker is provided between the second adder and the feedforward processing unit
in a frequency band lower than the first frequency. A first filter having a filter coefficient
exhibiting a characteristic of inclination at a slope of -6 dB / oct or less, and a gain of an
electrical signal to be input to the speaker less than a second frequency provided in a stage
preceding the feedforward processing unit And a second filter having a filter coefficient
exhibiting a characteristic inclined at a slope of 6 dB / oct or more in a frequency band, wherein
the first and second frequencies are open loop transmissions of a feedback loop formed by the
feedback processing unit. The frequency is higher than the gain crossover frequency indicated by
the characteristics.
[0024]
According to a sixteenth invention, in the twelfth invention, the filter coefficient in the
feedforward processing unit is a coefficient based on an intrinsic parameter of the speaker, and
the feedforward processing unit cancels the non-linear component of the parameter. And
processing the electrical signal output from the adder.
[0025]
In a seventeenth invention according to the twelfth invention, the filter coefficient in the
feedforward processing unit is a coefficient based on a parameter unique to the speaker, and the
parameter is a parameter that changes according to the vibration displacement of the speaker. It
features.
[0026]
In an eighteenth aspect based on the seventeenth aspect, the feedforward processing unit
receives the electrical signal output from the second adder as an input, and generates non-linear
distortion generated from the speaker based on a preset filter coefficient. And an electrical signal
output from the second adder as input, and a linear signal generating an electrical signal
indicating vibration displacement when assuming that the speaker vibrates linearly. And a filter,
08-05-2019
7
wherein the removal filter is characterized by referring to an electrical signal indicating vibration
displacement generated in the linear filter.
[0027]
In a nineteenth aspect based on the eighteenth aspect, the filter further includes: an amplification
section provided between the feedforward processing section and the speaker for amplifying the
gain of the electrical signal to be input to the speaker; The filter coefficient in the predetermined
characteristic conversion filter and the filter coefficient in the linear filter are filter coefficients
multiplied by the reciprocal of the gain amplified in the amplification unit.
[0028]
In a twentieth aspect based on the seventeenth aspect, the electric signal detected by the sensor
is an electric signal indicating vibration displacement of the speaker, and the feedforward
processing unit is an electric signal indicating vibration displacement detected in the sensor. To
refer to.
[0029]
In a twenty-first aspect based on the twelfth aspect, the electric signal to be input to the speaker
is provided at the front stage of the second adder, and a predetermined frequency characteristic
is divided by a characteristic relating to vibration of the speaker. It further comprises a pre-filter
that performs processing based on the filter coefficients obtained.
[0030]
According to a twenty-second invention, in the above-mentioned twelfth invention, the invention
further includes limiting means for limiting the level of the electrical signal so that the electrical
signal above the predetermined level is not input to the speaker.
[0031]
According to a twenty-third invention, in the twelfth invention, the feedforward processing unit
further includes an amplification unit provided between the feedforward processing unit and the
speaker for amplifying the gain of the electric signal to be input to the speaker. The filter
coefficient and the filter coefficient in the predetermined characteristic conversion filter are filter
coefficients multiplied by the reciprocal of the gain amplified in the amplification unit.
[0032]
08-05-2019
8
A twenty-fourth invention is an integrated circuit, which is a feedforward process for feedforward
processing of an electric signal to be input to a speaker based on a preset filter coefficient so as
to remove non-linear distortion generated from the speaker. And a feedback processing unit that
detects vibration of the speaker and performs feedback processing on an electrical signal relating
to the vibration with respect to the electrical signal to be input to the speaker, and the feedback
processing unit performs non-linear distortion generated from the speaker The electric signal
related to the vibration is feedback-processed so that the frequency characteristic corresponding
to the vibration of the speaker becomes a predetermined frequency characteristic so as to be
removed.
[0033]
According to the first aspect of the invention, it is possible to remove most of the non-linear
distortion by feed forward processing based on preset filter coefficients.
Furthermore, the feedback processing can remove distortion that is robust against, for example,
the secular change of the stiffness of the support system in the speaker.
That is, according to the present invention, the feedforward processing unit performs processing
based on the preset filter coefficient, and the feedback processing unit performs processing for
updating the speaker parameters by performing the above-mentioned robust distortion removal.
Thus, it is possible to provide a speaker device capable of more stable and highly feasible
distortion removal processing.
Furthermore, according to the present invention, it is possible to approximate the frequency
characteristic related to the vibration of the speaker to a predetermined frequency characteristic
by feedback processing.
[0034]
According to the second aspect of the invention, most of the non-linear distortion can be
removed by feed forward processing based on preset filter coefficients, and by feedback
processing based on an error signal, for example, stiffness of a support system in a speaker It is
possible to perform robust distortion removal against aging and the like.
08-05-2019
9
As a result, it is possible to provide a speaker device capable of more stable and highly feasible
distortion removal processing.
Furthermore, according to the present invention, it is possible to make the frequency
characteristic related to the vibration of the speaker close to the predetermined frequency
characteristic by the predetermined characteristic conversion filter.
[0035]
According to the third aspect of the present invention, non-linear distortion generated from the
speaker can be more effectively removed by processing the electric signal to be input to the
speaker so as to cancel the non-linear component of the parameter.
[0036]
According to the fourth aspect of the invention, it is possible to perform high-precision distortion
removal processing according to the vibration displacement of the speaker.
[0037]
According to the fifth aspect of the invention, processing based on vibration displacement when
the speaker vibrates linearly can be performed, and strain removal processing can be performed
more efficiently.
[0038]
According to the sixth aspect of the invention, even when the voltage that can be processed in
the internal calculation of the removal filter, the predetermined characteristic conversion filter,
and the linear filter is small, processing that maintains the distortion removal effect is possible.
Further, by providing the amplification unit in the feedback loop, the feedback gain is increased,
and the distortion reduction effect can be improved.
[0039]
08-05-2019
10
According to the seventh aspect, it is possible to perform distortion removal processing in line
with the actual vibration of the speaker.
[0040]
According to the eighth aspect of the present invention, in the characteristic relating to the
vibration output from the speaker, the convergence to the predetermined frequency
characteristic can be enhanced.
[0041]
According to the ninth aspect, breakage of the speaker due to excessive input can be prevented.
[0042]
According to the tenth aspect of the invention, even when the voltage that can be processed in
the internal calculation in the feedforward processing unit and the predetermined characteristic
conversion filter is small, it is possible to perform processing that maintains the distortion
removal effect.
Further, by providing the amplification unit in the feedback loop, the feedback gain is increased,
and the distortion reduction effect can be improved.
[0043]
According to the eleventh aspect, by arranging the feedforward processing unit in the feedback
loop, the distortion removal effect can be exhibited to a lower frequency band even when the
amplitude of the speaker is increased.
[0044]
According to the twelfth aspect of the present invention, by arranging the feedforward
processing unit in the feedback loop, the distortion removal effect can be exhibited to a lower
frequency band even if the amplitude of the speaker increases.
[0045]
According to the thirteenth aspect, since the gain crossover frequency is lowered by the first
08-05-2019
11
filter, the distortion removing effect can be exhibited to a lower frequency band.
[0046]
According to the fourteenth aspect of the present invention, since the second filter does not
receive an electrical signal lower than the gain crossover frequency, distortion caused by the
electrical signal lower than the gain crossover frequency can be eliminated in advance. A high
distortion removal effect can be obtained.
[0047]
According to the fifteenth invention, the gain crossover frequency is lowered by the first filter, so
that the distortion removing effect can be exhibited to a lower frequency band.
Furthermore, since the second filter does not receive an electrical signal at or below the gain
crossover frequency, distortion caused by the input of the electrical signal at or below the gain
crossover frequency can be removed in advance to obtain a higher distortion removal effect. Can.
[0048]
Hereinafter, embodiments of the present invention will be described with reference to the
drawings.
[0049]
First Embodiment A speaker device 1 according to a first embodiment of the present invention
will be described with reference to FIG.
FIG. 1 is a block diagram showing a configuration example of the speaker device 1 according to
the first embodiment.
In FIG. 1, the speaker device 1 includes a non-linear component removal filter 10, a linear filter
11, an ideal filter 12, adders 13 and 14, a feedback control filter 15, a speaker 16, and a sensor
08-05-2019
12
17.
[0050]
Here, first, with reference to FIG. 2, the cause of generation of non-linear distortion in the
speaker 16 will be described.
FIG. 2 is a cross-sectional view of a general speaker 16.
In FIG. 2, the speaker 16 includes a voice coil 161, a diaphragm 162, a magnet 163, a magnetic
circuit 164, a damper 166, and an edge 167.
A magnetic gap 165 is formed in the magnetic circuit 164 shown in FIG.
Then, the voice coil 161 vibrates integrally in the vibration displacement x-axis direction with the
diaphragm 162 according to Fleming's left-hand rule with the magnetic flux density B in the
magnetic gap 165 and the current flowing through the voice coil 161.
The diaphragm 162 is supported by the damper 166 and the edge 167 to stably vibrate in the
vibration displacement x-axis direction to emit a sound.
In addition, the speaker 16 shown in FIG. 2 is an example, and is not limited to this.
For example, it may be a magnetic shield type speaker including a cancel magnet, or may be a
speaker constituting an internal magnetic type magnetic circuit.
Further, in FIG. 2, the position where the vibration displacement x is 0 indicates the central
position where the voice coil 161 and the diaphragm 162 vibrate, and corresponds to the origin
where the vibration displacement x shown in FIGS. Do.
08-05-2019
13
[0051]
In the speaker 16, three main causes can be mentioned as the causes of the non-linear distortion.
The first factor relates to the magnetic flux density B generated in the magnetic gap 165.
FIG. 3 is a view showing an example of the characteristic of the force coefficient Bl with respect
to the vibration displacement x in the vicinity of the magnetic gap 165. As shown in FIG.
When the amplitude of the voice coil 161 is small, that is, when the absolute value of the
vibration displacement x is small (in the vicinity of x = 0), the magnetic flux density B is
substantially constant.
However, when the amplitude of the voice coil 161 is large, that is, when the absolute value of
the vibration displacement x is large, the magnetic flux density B rapidly decreases.
This is because in the magnetic circuit 164, the magnetic path is less likely to be formed as it
moves away from the vicinity of the center of the magnetic gap 165 (x = 0) in the vibration
displacement x-axis direction.
Therefore, the relationship between the force coefficient Bl obtained by the magnetic flux density
B and the vibration displacement x of the voice coil 161 is as shown in FIG.
The characteristics of the force coefficient Bl shown in FIG. 3 change in accordance with the
vibration displacement x, and are expressed as a function Bl (x) of the vibration displacement x.
[0052]
Here, the driving force F (t) for vibrating the voice coil 161 is expressed by the following
equation (1), where I (t) is the current of the input signal flowing through the voice coil 161.
08-05-2019
14
F (t) = B1 (x) * I (t) (1) As shown in FIG. 3, when the amplitude of the voice coil 161 increases, the
value of the force coefficient B1 (x) decreases.
Therefore, according to the above equation (1), when the amplitude increases, the driving force F
(t) is not proportional to the level of the input signal I (t).
Needless to say, the vibration displacement x is not proportional to the level of the input signal I
(t) unless the driving force F (t) is proportional to the level of the input signal I (t).
As a result, non-linear distortion occurs from the speaker 16.
[0053]
The second factor relates to the support system such as the damper 166 and the edge 167.
The damper 166 and the edge 167 do not extend infinitely because of their shapes, and begin to
be stretched when they extend to some extent.
FIG. 4 is a diagram showing an example of the characteristic K of the stiffness of the support
system with respect to the vibration displacement x.
In FIG. 4, when the amplitude of the voice coil 161 is small, that is, when the absolute value of
the vibration displacement x is small, the stiffness K is substantially constant.
However, when the amplitude of the voice coil 161 is large, that is, when the absolute value of
the vibration displacement x is large, the value of the stiffness K becomes large.
Thus, as the amplitude increases, the value of the stiffness K changes, and the vibration
displacement x is not proportional to the driving force F (t).
08-05-2019
15
Further, if the vibration displacement x is not proportional to the driving force F (t), from the
above equation (1), the vibration displacement x is not proportional to the level of the input
signal I (t).
As a result, non-linear distortion occurs from the speaker 16.
[0054]
FIG. 5 is a diagram showing a change in the characteristic of stiffness K with respect to input
signal I (t). As shown in FIG. 5, the property of stiffness K changes according to the level of I (t)
and does not always become a constant curve. Further, since the damper 166 and the edge 167
are made of a material such as cloth or resin, the characteristics of the stiffness K shown in FIG. 4
change also due to the secular change of the material and the creep phenomenon. Due to these
factors as well, the vibration displacement x is not proportional to the level of the input signal I
(t), and non-linear distortion occurs from the speaker 16.
[0055]
The third factor relates to the electrical impedance characteristic of the voice coil 161. Generally,
high magnetic permeability materials such as iron are used for the speaker's magnetic circuit. For
this reason, the inductance component of the voice coil 161 changes depending on the
magnitude of the amplitude. Also, the voice coil 161 generates heat when an electrical signal is
input. Thereby, the resistance component of the voice coil 161 changes with time. Due to these
factors, the current flowing through the voice coil 161 is distorted, and non-linear distortion
occurs from the speaker 16. Non-linear distortion occurs in the speaker 16 due to the three main
factors as described above.
[0056]
When the speaker 16 is driven at a constant voltage, the relationship between the voltage E (t) of
the input signal input to the speaker 16 and the vibration displacement x (t) is generally
expressed by the following expression (2) . Bl * E (t) / Ze = K * x (t) + (r + Bl <2> / Ze) * dx (t) / dt
08-05-2019
16
+ m * d <2> x (t) / dt <2> ... (2) However, in Equation (2), the stiffness of the support system is K,
the mechanical resistance of the speaker 16 is r, the electrical impedance of the voice coil 161 is
Ze, and the mass of the vibration system is m.
[0057]
Here, among the above three factors, the non-linear distortion generated in the low frequency
band is particularly influenced by the parameters of the force coefficient Bl and the stiffness K.
Therefore, in the above equation (2), the force coefficient Bl and the stiffness K shown in FIGS. 3
and 4 can be expressed as a function of the vibration displacement x as the following equation
(3). Bl (x) * E (t) / Ze = K (x) * x (t) + (r + Bl (x) <2> / Ze) * dx (t) / dt + m * d <2> x (t) / dt <2> (3)
Further, when Bl (x) and K (x) are polynomially approximated for the vibration displacement x
and modeled, equations (4) and (5) are obtained. Bl (x) = A0 + A1 * x + A2 * x <2> + A3 * x <3> +
(4) K (x) = K0 + K1 * x + K2 * x <2> + K3 * x <3> + (5) In the above equations (4) and (5), A0 and
K0 are parameters of linear components that do not depend on the vibration displacement x.
Therefore, when Expression (4) and Expression (5) are divided into linear components and nonlinear components, they are expressed as Expression (6) and Expression (7), respectively. Bl (x) =
A0 + Ax (6) K (x) = K0 + Kx (7) where Ax is a nonlinear component of Bl (x) and Kx is a nonlinear
component of K (x) . Therefore, when Formula (6) and Formula (7) are substituted for Bl (x) and K
(x) in Formula (3), Formula (8) is obtained. (A0 + Ax) * E (t) / Ze = (K0 + Kx) * x (t) + [r + (A0 + Ax)
<2> / Ze] * dx (t) / dt + m * d <2 > x (t) / dt <2> (8)
[0058]
Next, an operation process of the speaker device 1 shown in FIG. 1 will be described. In the
speaker device 1 according to the present embodiment, roughly, feedforward processing by the
non-linear component removal filter 10 and the linear filter 11, the ideal filter 12, the sensor 17,
the adder 14, the feedback control filter 15, and the adder 13 And feedback processing is
performed. Thus, the non-linear component removal filter 10 and the linear filter 11 correspond
to the feedforward processing unit of the present invention. Further, the ideal filter 12, the
sensor 17, the adder 14, the feedback control filter 15, and the adder 13 correspond to the
feedback processing of the present invention.
[0059]
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17
First, feedforward processing by the non-linear component removal filter 10 and the linear filter
11 will be described. An electrical signal is input as an input signal to the non-linear component
removal filter 10, the linear filter 11, and the ideal filter 12, respectively. The processing of the
ideal filter 12 will be described later.
[0060]
The non-linear component removal filter 10 cancels the non-linear component of the modeled
parameter based on a predetermined filter coefficient obtained with reference to the vibration
displacement x (t) in pseudo linear operation generated in the linear filter 11 To process the
input signal. Then, the signal processed by the non-linear component removal filter 10 is output
to the adder 13. Hereinafter, predetermined filter coefficients set in the non-linear component
removal filter 10 will be described.
[0061]
The operation equation of the speaker 16 is as shown by the above equation (8). From the above
equation (8), the operation equation that does not include the nonlinear components (B1x and
Kx) of the parameter, that is, the operation equation in the linear operation where non-linear
distortion does not occur is the following equation (9). A0 * E (t) / Ze = K0 * x (t) + [r + A0 <2> /
Ze] * dx (t) / dt + m * d <2> x (t) / dt <2> ... (9) Therefore, by subtracting the equation (9) from
the equation (8), it is possible to extract an operational equation of only the nonlinear component
of the speaker as the equation (10). Ax * E (t) / Ze = Kx * x (t) + [(2 * A0 * Ax + A0 <2>) / Ze] * dx
(t) / dt (10) Further, from Expression (8) By reducing the equation (10), it is possible to obtain an
operation equation from which the non-linear component is removed as the equation (11). (A0 +
Ax) * E (t) / Ze-Ax * E (t) / Ze = (K0 + Kx) * x (t) + [r + (A0 + Ax) <2> / Ze] * dx (t ) / dt + m * d <2>
x (t) / dt <2> -Kx * x (t) + [(2 * A0 * Ax + A0 <2>) / Ze] * dx (t) / dt (11) Here, if the right side of
the equation (11) is made equal to the right side of the equation (8) which is the operation
equation of the speaker 16 originally, the equation (11) is expressed as the equation (12). (A0 +
Ax) * E (t) / Ze-Ax * E (t) / Ze + Kx * x (t) + [(2 * A0 * Ax + A0 <2>) / Ze] * dx (t) / dt = (K0 + Kx) * x
(t) + [r + (A0 + Ax) <2> / Ze] * dx (t) / dt + m * d <2> x (t) / dt <2> (12) The following equation
(13) can be obtained by rearranging the left side of the above equation (12). The left side of
equation (13) is a filter coefficient for canceling the non-linear component of the parameter. (A0
+ Ax) / Ze * [E (t) -Ze / (A0 + Ax) * (Ax / Ze * E (t)-(2 * A0 * Ax + Ax <2>)] / Ze * dx (t ) / dt-Kx * x
(t)] = (K0 + Kx) * x (t) + [r + (A0 + Ax) <2> / Ze] * dx (t) / dt + m * d <2 > x (t) / dt <2> (13)
08-05-2019
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[0062]
In the above filter coefficients, the parameters A0 and Ax related to the above-described force
coefficient Bl, the parameters K0 and Kx related to the stiffness K, and the electrical impedance
Ze are inherent parameters of the connected speaker 16. It is a preset parameter that constitutes
the filter coefficient. Further, from the left side of Equation (13), it can be seen that the value of
vibration displacement x (t) is also required as a parameter necessary for the filter coefficient of
the non-linear component removal filter 10. The vibration displacement x (t) is generated in the
linear filter 11 described below.
[0063]
The linear filter 11 generates vibration displacement x (t) when assuming that the speaker 16
linearly operates from the input signal based on a preset filter coefficient. That is, the linear filter
11 generates vibration displacement x (t) at the time of pseudo linear operation. As described
above, the equation for the linear operation of the speaker 16 is as shown in equation (9).
Therefore, Laplace transform of equation (9) to obtain a transfer function yields equation (14).
The right side of equation (14) is the filter coefficient of the linear filter 11. Here, x (s) is a
transfer function of the vibration displacement x (t), and E (s) is a transfer function of the voltage
of the input signal. x (s) / E (s) = (A0 / Ze) / [K0 + s * (r + A0 <2> / Ze) + s <2> * m] (14)
[0064]
Thus, the feedforward processing by the non-linear component removal filter 10 and the linear
filter 11 cancels out the non-linear components of the modeled force coefficient B1 (x) and
stiffness K (x) as shown in the above equation (8). . Thereby, non-linear distortion resulting from
the non-linear component can be removed. Also, this feedforward processing cancels out nonlinear components so that the speaker 16 operates in a linear manner. Since the non-linear
component removal filter 10 refers to the vibration displacement x (t) at the time of linear
operation of the speaker 16, a more efficient distortion removal effect can be obtained.
[0065]
Next, feedback processing in the ideal filter 12, the sensor 17, the adder 14, the feedback control
filter 15, and the adder 13 will be described.
08-05-2019
19
[0066]
The ideal filter 12 is a filter that uses the transfer function F (s) of the desired output
characteristic as a filter coefficient when the characteristic corresponding to the vibration of the
speaker 16 (hereinafter referred to as the output characteristic) is the desired output
characteristic. is there.
That is, the ideal filter 12 is a filter that converts the frequency characteristic of the input signal
into a desired output characteristic. Here, the signal converted into the desired output
characteristic is taken as the desired characteristic signal f (t). The desired characteristic signal f
(t) is output to the adder 14. The output characteristics of the speaker 16 include various
characteristics such as vibration displacement characteristics, speed characteristics, and
acceleration characteristics (sound pressure characteristics). For example, as shown in FIG. 6, it is
assumed that the sound pressure frequency characteristic (acceleration characteristic) of the
actual speaker 16 is a characteristic shown in A of FIG. FIG. 6 is a diagram showing a desired
output characteristic set as the filter coefficient of the ideal filter 12. In FIG. 6, when the sound
pressure frequency characteristic of the speaker 16 is broadened into a flat characteristic like the
characteristic shown by B, the transfer function F (s) of the characteristic shown by B is a filter of
the ideal filter 12 It may be set as a coefficient.
[0067]
The sensor 17 detects the vibration of the speaker 16 and outputs a detection signal y (t) having
an output characteristic of the speaker 16. The detection signal y (t) output from the sensor 17 is
appropriately amplified and output to the adder 14. The sensor 17 is, for example, a microphone,
a laser displacement meter, an acceleration pickup, or the like. Here, the type of signal
characteristic output to the adder 14 is the same type as the output characteristic of the desired
characteristic signal f (t) described above. That is, in the ideal filter 12, when the output
characteristic of the desired characteristic signal f (t) is, for example, the vibration displacement
characteristic of the speaker 16, the signal output to the adder 14 is the signal of the vibration
displacement characteristic. In this case, the sensor 17 may use a sensor that detects the
vibration of the speaker 16 and outputs the vibration displacement. Alternatively, even if a
sensor that outputs the speed characteristic and the acceleration characteristic of the speaker 16
is used as the sensor 17, a differential circuit or an integration circuit is appropriately provided
between the sensor 17 and the adder 14, and a signal output to the adder 14 The type of the
characteristic of the above may be converted into the vibration displacement characteristic.
08-05-2019
20
[0068]
The sound pressure frequency characteristic of the speaker is a characteristic proportional to the
acceleration characteristic. Therefore, the characteristic of the desired characteristic signal f (t)
output from the ideal filter 12 indicates the acceleration characteristic of the speaker 16, and the
sensor 17 is an acceleration pickup and the characteristic of the signal output from the sensor
17 indicates the acceleration characteristic. When shown, the distortion removal effect is the
highest.
[0069]
Hereinafter, for the purpose of explanation, it is assumed that the type of the characteristic of the
detection signal y (t) output from the sensor 17 is the same as the output characteristic of the
desired characteristic signal f (t) output from the ideal filter 12. That is, the case where it is not
necessary to provide a differentiating circuit or an integrating circuit between the sensor 17 and
the adder 14 is considered.
[0070]
The adder 14 subtracts the detection signal y (t) output from the sensor 17 from the desired
characteristic signal f (t) output from the ideal filter 12 and subtracts the signal (f (t) -y (t) Is
output to the feedback control filter 15 as an error signal e (t). The error signal e (t) is
appropriately adjusted in gain and the like in the feedback control filter 15, and is fed back to the
adder 13. Then, the adder 13 adds the output signal of the non-linear component removal filter
10 and the error signal e (t) output from the feedback control filter 15 and outputs the result to
the speaker 16. The feedback control filter 15 is basically a filter for adjusting the gain, that is,
an amplifier, and the larger the gain, the larger the distortion removal effect.
[0071]
Here, as described above, the stiffness K of the support system changes with time. Further, as
shown in FIG. 5, the characteristics of the stiffness K also change depending on the size of the
08-05-2019
21
input. And in this case, the output characteristic of the speaker 16 also changes. On the other
hand, the sensor 17 detects this changed output characteristic of the speaker 16, and the abovementioned error signal e (t) is detected signal y (t) outputted from the sensor 17 and the desired
characteristic signal r (t). And the difference signal. Therefore, the secular change of the stiffness
K and the characteristic change due to the size of the input are reflected in the error signal e (t).
Then, the error signal e (t) is fed back to the adder 13 via the feedback control filter 15, whereby
the aging of the stiffness K and the characteristic change due to the size of the input are
cancelled.
[0072]
Thus, the feedback processing in the ideal filter 12, the sensor 17, the adder 14, the feedback
control filter 15, and the adder 13 is robust against the aging of the stiffness K of the support
system and the characteristic change due to the size of the input. Distortion removal processing
can be performed.
[0073]
The error signal e (t) also includes the change in the electrical impedance characteristic of the
voice coil 161 (particularly, the change due to heat generation), which is the cause of the third
non-linear distortion described above.
Therefore, non-linear distortion due to the change can also be removed by the above feedback
processing.
[0074]
Further, in generating the error signal e (t), the ideal filter 12 uses a signal f (t) having a desired
output characteristic (transfer function F (s)). Then, the error signal e (t) is subjected to feedback
processing, so that the actual output characteristic of the speaker 16 can be made closer to the
desired output characteristic.
[0075]
08-05-2019
22
As described above, according to the speaker device 1 according to the present embodiment,
non-linear distortion of most of the speakers can be removed by feed forward processing, and
aging of stiffness of the support system or magnitude of input by feedback processing. Robust
distortion removal processing can be performed on characteristic changes due to the As a result,
there is no need for an adaptive parameter updating circuit that requires complicated and
massive calculations, and it is possible to prevent cost increase, and it is possible to provide a
speaker device capable of more stable and highly feasible distortion removal processing.
[0076]
The above-described feedback control filter 15 may have characteristics such as a low pass filter
as well as gain adjustment. For example, there is a possibility that oscillation may occur if the
middle to high frequency characteristic of the speaker 16 is largely disturbed and the error
signal e (t) is fed back as it is. At this time, oscillation can be prevented by providing the
characteristics of the low pass filter in the feedback control filter 15 and cutting the middle and
high frequency components. Further, in the speaker device 1 shown in FIG. 1, the feedback
control filter 15 may be omitted if there is no fear of oscillation due to the error signal e (t) or the
need for gain adjustment.
[0077]
Further, in the non-linear component removal filter 10 described above, non-linear distortion
caused by the force coefficient B1 and the stiffness K of the support system is removed by using
the filter coefficient shown in the equation (13) derived from the equation (8). However, it is not
limited to this. In Equation (8), the above-described electrical impedance characteristic Ze of the
voice coil 161 is reflected as a function Ze (x) of the vibration displacement x, and a filter
coefficient in consideration of the electrical impedance characteristic Ze is set from Equation
(14). May be Thereby, in the feedforward processing in the non-linear component removal filter
10 and the linear filter 11, non-linear distortion due to the fluctuation based on the vibration
displacement x (t) of the electrical impedance characteristic Ze can be removed.
[0078]
Also, in the non-linear component removal filter 10 described above, the vibration displacement
x (t) in linear operation generated pseudo by the linear filter 11 is referred to, but as shown in
08-05-2019
23
FIG. It may be a reference. That is, by directly referring to the output of the sensor 17, the linear
filter 11 can be omitted. Further, in this case, the vibration displacement x (t) is the vibration
displacement x (t) of the actual speaker, and the non-linear component removal filter 10 can
perform processing in line with the vibration displacement of the actual speaker. FIG. 7 is a block
diagram showing a configuration example of the speaker device 1 when the non-linear
component removal filter 10 refers to the output signal of the sensor 17. At this time, since the
signal referred to by the non-linear component removal filter 10 is the vibration displacement x
(t), the sensor 17 may be any one that detects the vibration displacement characteristic of the
speaker 16. Further, even if the signal detected by the sensor 17 itself is a velocity characteristic
or an acceleration characteristic, it is possible to obtain a vibration displacement characteristic by
appropriately using a differentiation circuit and an integration circuit.
[0079]
Second Embodiment A speaker device 2 according to a second embodiment of the present
invention will be described with reference to FIG. FIG. 8 is a block diagram showing a
configuration example of the speaker device 2 according to the second embodiment. In FIG. 8,
the speaker device 2 includes a non-linear component removal filter 10, a linear filter 11, an
ideal filter 12, an adder 13, an adder 14, a feedback control filter 15, a speaker 16, a sensor 17,
and a pre-filter 20. As shown in FIG. 8, the speaker device 2 according to the present
embodiment is different from the above-described speaker device 1 shown in FIG. 1 in that a
front-stage filter 20 is newly provided. Hereinafter, differences will be mainly described. Also, the
non-linear component removal filter 10, the linear filter 11, the ideal filter 12, the adder 13, the
adder 14, the feedback control filter 15, the speaker 16, and the sensor 17 are the same as the
respective configurations described in the first embodiment. Since there is the same code, the
description will be omitted.
[0080]
The pre-stage filter 20 is located in front of the non-linear component removal filter 10 and the
linear filter 11, and processes the input signal based on a predetermined filter coefficient, using
the electrical signal as the input signal. The signal processed by the pre-filter 20 is input to the
non-linear component removal filter 10 and the linear filter 11, respectively. Here, the filter
coefficient of the pre-filter 20 is the transfer function F (s) of the desired output characteristic
that is the filter coefficient of the ideal filter 12, the transfer function P (s) of the output
characteristic during linear operation of the actual speaker 16. It is F (s) / P (s) divided by). The
output characteristic of the transfer function P (s) is made the same as the type of the desired
08-05-2019
24
output characteristic of the ideal filter 12. That is, as described in the first embodiment, for
example, when the transfer function F (s) is based on the vibration displacement characteristic of
the speaker 16, the transfer function P (s) also causes the vibration displacement when the
speaker 16 operates in a linear manner. It is a function based on the property.
[0081]
Here, it is assumed that the transfer function of the input signal voltage input to the pre-filter 20
is E (s). At this time, the output signal of the pre-stage filter 20 is E (s) * F (s) / P (s). Then, when
output at the speaker 16 via the non-linear component removal filter 10, the transfer function P
(s) of the speaker 16 is multiplied, so the output characteristic of the speaker 16 is finally E (s) *
F ( s). That is, the output characteristic of the speaker 16 converges to the target characteristic F
(s). At this time, the transfer function of the detection signal y (t) output from the sensor 17 is E
(s) * F (s). Further, an input signal to be a transfer function E (s) is input to the ideal filter 12. At
this time, since the filter coefficient of the ideal filter 12 is F (s), the transfer function of the
output signal f (t) of the ideal filter 12 is E (s) * F (s). Then, the adder 14 subtracts the detection
signal y (t) from the output signal f (t) from the ideal filter 12. At this time, transfer functions of
the output signal f (t) and the detection signal y (t) are both equal to E (s) * F (s), and the error
signal e (t) is zero.
[0082]
Further, for example, it is assumed that the transfer function of the speaker fluctuates from P (s)
to P '(s) due to the secular change of the stiffness K of the support system. At this time, the
transfer function Y (s) / E (s) of the entire speaker device 2 shown in FIG. 8 is expressed by
equation (15). Y (s) is obtained by Laplace transform of the output signal y (t) from the speaker
16. E (s) is the Laplace transform of the input signal voltage. Y (s) / E (s) = (P '(s) * [1 + P (s)] / (P
(s) * [1 + P' (s)]) * F (s) (( 15) From the above equation (15), when the transfer function P (s) of
the speaker 16 does not change (when P ′ (s) = P (s)), the right side of the equation (15) is F (s)
Become. That is, the output characteristic of the speaker 16 converges to the desired
characteristic F (s).
[0083]
Next, in the speaker device 1 shown in FIG. 1 which does not have the pre-filter 20, assuming
08-05-2019
25
that the transfer function when the speaker 16 linearly operates is P (s), the entire speaker
device 1 shown in FIG. The transfer function Y (s) / E (s) of the equation becomes equation (16).
Y (s) / E (s) = (P (s) * [1 + F (s)]) / [1 + P (s)] (16) From the above equation (16), the transfer
function of the speaker 16 When P (s) does not change (when P ′ (s) = P (s)), the right side of
equation (16) does not become F (s). That is, the output characteristic of the speaker 16 does not
converge to the desired characteristic F (s).
[0084]
Further, assuming that the transfer function of the speaker 16 changes from P (s) to P '(s), the
transfer function Y (s) / E (s) of the speaker device 1 shown in FIG. Become. Y (s) / E (s) = (P '(s) *
[1 + F (s)]) / [1 + P' (s)] (17)
[0085]
As described above, in the speaker device 1 shown in FIG. 1, as shown in the equations (16) and
(17), the characteristic that the output characteristic of the speaker 16 approaches F (s) by
providing the ideal filter 12 However, the desired characteristic F (s) does not converge
regardless of the variation of the transfer function of the speaker 16. On the other hand, in the
speaker device 2 shown in FIG. 8, by providing the pre-filter 20, at least the transfer function of
the speaker converges to F (s) when it does not change. That is, the pre-filter 20 plays a role in
enhancing the convergence of the speaker 16 to the desired output characteristic.
[0086]
As described above, in the speaker device 2 according to the present embodiment, by providing
the pre-stage filter 20, the convergence to the desired output characteristic (transfer function F
(s)) can be made extremely high.
[0087]
The above-described feedback control filter 15 may have characteristics such as a low pass filter
as well as the gain adjustment, as in the first embodiment.
08-05-2019
26
Further, in the speaker device 2 shown in FIG. 8, the feedback control filter 15 may be omitted if
there is no fear of oscillation due to the error signal e (t) or the necessity of gain adjustment.
[0088]
Further, in the non-linear component removal filter 10 described above, as in the first
embodiment, the force coefficient Bl and the stiffness K of the support system are obtained by
using the filter coefficient shown in the equation (13) derived from the equation (8). It is said that
non-linear distortion due to is removed, but it is not limited to this. In Equation (8), the abovedescribed electrical impedance characteristic Ze of the voice coil 161 is reflected as a function Ze
(x) of the vibration displacement x, and a filter coefficient in consideration of the electrical
impedance characteristic Ze is set from Equation (14). May be
[0089]
Moreover, although the structure which connected the input of the linear filter 11 and the output
of the front | former stage filter 20 was shown in FIG. 8 mentioned above, it is not limited to this.
Even if the input of the linear filter 11 is the same as the input of the pre-stage filter 20 and the
ideal filter 12 as shown in FIG. 9, the same effect as the effect obtained by the configuration
shown in FIG. it can. FIG. 9 is a block diagram showing a configuration example in which the
input of the linear filter 11 shown in FIG. 8 is changed.
[0090]
Further, in the non-linear component removal filter 10 described above, as in the first
embodiment, the vibration displacement x (t) in linear operation generated pseudo by the linear
filter 11 is referred to, but as shown in FIG. Alternatively, the output signal of the sensor 17 may
be directly referenced. That is, by directly referring to the output of the sensor 17, the linear
filter 11 can be omitted. FIG. 10 is a block diagram showing a configuration example of the
speaker device 2 when the non-linear component removal filter 10 refers to the output signal of
the sensor 17. At this time, since the signal referred to by the non-linear component removal
filter 10 is the vibration displacement x (t), the sensor 17 may be any one that detects the
vibration displacement characteristic of the speaker 16. Further, even if the signal detected by
the sensor 17 itself is a velocity characteristic or an acceleration characteristic, it is possible to
obtain a vibration displacement characteristic by appropriately using a differentiation circuit and
08-05-2019
27
an integration circuit.
[0091]
Third Embodiment A loudspeaker device 3 according to a third embodiment of the present
invention will be described with reference to FIG. FIG. 11 is a block diagram showing a
configuration example of the speaker device 3 according to the third embodiment. In FIG. 11, the
speaker device 3 includes a non-linear component removal filter 10, an ideal filter 12, an adder
13, an adder 14, a feedback control filter 15, a speaker 16, a sensor 17, and a pre-filter 20. The
speaker device 3 according to this embodiment is different from the speaker devices 1 and 2
shown in FIG. 1 and FIGS. 7 to 10 in that the non-linear component removal filter 10 is disposed
between the adder 13 and the speaker 16. The speaker device is capable of extending the
frequency band where the distortion removal effect can be obtained due to the difference to a
low frequency range.
[0092]
Hereinafter, with reference to FIG. 11, the above-mentioned difference will be mainly described.
FIG. 11 shows a configuration example in which the arrangement position of the non-linear
component removal filter 10 is changed with respect to the speaker device 2 shown in FIG. 10 as
the speaker device 3. In FIG. 11, although the codes relating to the input and output of the
adders 13 and 14 are different from the codes shown in FIG. 10, the operation and the effect are
the same regardless of which code if the phase relationship is equal. . Further, the non-linear
component removal filter 10, the ideal filter 12, the adder 13, the adder 14, the feedback control
filter 15, the speaker 16, the sensor 17, and the pre-stage filter 20 have the respective
configurations described in the first and second embodiments. The same reference numerals are
assigned and the description is omitted.
[0093]
The non-linear component removal filter 10 is disposed between the adder 13 and the speaker
16. That is, the non-linear component removal filter 10 is disposed in a feedback loop formed by
the sensor 17, the adder 14, the feedback control filter 15, the adder 13, and the speaker 16. In
this case, a combination of the non-linear component removal filter 10 and the speaker 16 can
be considered as a control object in linear two-degree-of-freedom control.
08-05-2019
28
[0094]
Here, as described in the first embodiment, the non-linear component removal filter 10 cancels
out the non-linear component of the modeled stiffness K to remove non-linear distortion
generated from the speaker 16. Therefore, it can be considered that the above-mentioned
controlled object is one in which the non-linear distortion of the speaker 16 is removed to some
extent by the non-linear component removal filter 10. By arranging such a control target in the
feedback loop, the change of the stiffness K with respect to the vibration displacement X shown
in FIG. 4 is reduced in the feedback loop. That is, even if the amplitude of the speaker 16
increases, the stiffness K does not change much. Further, since the change in the stiffness K is
reduced, the change in the lowest resonance frequency f0 of the speaker 16 is also reduced.
[0095]
On the other hand, in the speaker device 2 shown in FIG. 10, the non-linear component removal
filter 10 is not disposed in the feedback loop. Therefore, in the speaker device 2 shown in FIG.
10, the control target is the speaker 16 alone, and the above-mentioned non-linear distortion is
not removed to some extent in the feedback loop.
[0096]
As described above, when attention is paid to the processing in the feedback loop, in the speaker
device 3 according to the present embodiment, the change of the minimum resonance frequency
f0 of the speaker 16 is smaller than that of the speaker device 2 shown in FIG.
[0097]
Next, with reference to the gain characteristics G1 to G4 and the phase characteristic P of the
speaker device 3 shown in FIG.
FIG. 12 is a diagram showing gain characteristics and phase characteristics of the speaker device
3. The gain characteristics G1 to G4 shown in FIG. 12 are open loop transfer characteristics.
Further, a gain characteristic G1 indicated by a solid line in FIG. 12 indicates a sound pressure
08-05-2019
29
frequency characteristic of the speaker 16, that is, a characteristic proportional to an
acceleration characteristic. The gain characteristics G2 to G4 indicated by dotted lines will be
described later.
[0098]
According to the gain characteristic G1, it is understood that the gain is attenuated at a slope of
−12 dB / oct in the frequency band lower than the lowest resonance frequency f0. According to
the phase characteristic P shown in FIG. 12, it can be seen that the phase is shifted by 90 ° at
the lowest resonance frequency f0. Also, it can be seen that the phase shift approaches 180 ° as
the frequency is lower at the lowest resonance frequency f0 or lower. Also, it can be seen that the
phase shift approaches 0 ° as the frequency is higher at the lowest resonance frequency f0 or
higher.
[0099]
Here, it is assumed that the gain of the error signal e (t) input to the adder 13 is adjusted in the
feedback control filter 15 shown in FIG. In this case, the gain characteristic G1 changes to the
gain characteristic G2, G3 or G4 shown by the dotted line in FIG. 12 according to the magnitude
of the gain adjusted in the feedback control filter 15. The magnitude of the input to the speaker
16 changes in accordance with the magnitude of the gain adjusted in the feedback control filter
15. Then, the magnitude of the amplitude of the speaker 16 changes as the magnitude of the
input to the speaker changes. Here, as described above, in the speaker device 3, even if the
amplitude of the speaker 16 increases, the change in the lowest resonance frequency f0 is small.
Therefore, the lowest resonance frequencies of the gain characteristics G2, G3 or G4 shown by
dotted lines in FIG. 12 are all close to f0.
[0100]
Next, consider the evaluation values of gain margin and phase margin. The gain margin indicates
how negative the gain of the open loop transfer characteristic takes when the phase of the open
loop characteristic is 180 °. The frequency at which the phase is 180 ° is called a phase
crossover frequency fpc. The phase margin indicates how negative the phase of the open loop
transfer characteristic is to 180 ° when the gain of the open loop transfer characteristic is 0 dB.
The frequency at which the gain is 0 dB is referred to as a gain crossover frequency fgc.
08-05-2019
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[0101]
Here, the frequency characteristic of the feedback loop of the speaker device 2 shown in FIG. 10
is analyzed. In the feedback loop of the speaker device 2 shown in FIG. 10, since the signal
indicating the normal acceleration characteristic is fed back, the frequency characteristic largely
changes and analysis becomes difficult. Therefore, in the analysis of the frequency characteristic,
it is considered to add the ideal filter 12 as shown in FIG. That is, the ideal filter 12 is added, and
analysis is performed in the state where the frequency characteristic does not change. FIG. 13 is
a diagram showing a configuration used for analysis of frequency characteristics of the speaker
device 2 shown in FIG.
[0102]
FIG. 14 shows the sound pressure frequency characteristic, the second-order distortion
characteristic, and the third-order distortion characteristic when the magnitude of the input to
the speaker 16 of FIG. 13 is changed. Specifically, as shown in FIG. 14, sound pressure frequency
characteristics, second-order distortion characteristics, and third-order distortion characteristics
are shown respectively when the input to the speaker 16 is 1V, 5W, 10W, 20W, and 40W. There
is. As can be seen from FIG. 14, as the input is increased, the level of second-order and thirdorder distortion is increased. This is because as the input increases, the stiffness increases and
the gain crossover frequency fgc increases. Thus, it can be said that the lower limit frequency of
the frequency band where the distortion removal effect is obtained is in proportion to the gain
crossover frequency fgc.
[0103]
Hereinafter, with reference to FIG. 12 again, the reason why the speaker device 3 can extend the
frequency band in which the distortion removal effect can be obtained to the low frequency
range will be described. In FIG. 12, when adjustment is performed to increase the gain in the
feedback control filter 15, the gain characteristic G1 becomes a characteristic indicated by the
gain characteristic G2. At this time, the gain crossover frequency fgc2 in the gain characteristic
G2 is a frequency smaller than the gain crossover frequency fgc1. This is because, as described
above, in the speaker device 3, even if the amplitude of the speaker 16 changes, the change in
the lowest resonance frequency f0 is small. As described above, in the speaker device 3, the
08-05-2019
31
frequency band in which the distortion removal effect is obtained is extended to the low band in
proportion to the gain crossover frequency fgc 2.
[0104]
On the other hand, in the speaker device 2 shown in FIG. 10, as described above, the non-linear
component removal filter 10 is not disposed in the feedback loop. Therefore, in the speaker
device 2 shown in FIG. 10, when the input to the speaker 16 becomes large, that is, when
adjustment is made to increase the gain in the feedback control filter 15, the gain characteristic
G1 becomes a characteristic shown by the gain characteristic G2 '. That is, the value of the
stiffness K increases and the lowest resonance frequency f0 rises to f0 ‘. Further, as the lowest
resonance frequency f0 rises, the gain crossover frequency also rises to the gain crossover
frequency fgc2 '. Therefore, in the speaker device 2, the frequency band in which the distortion
removal effect is obtained is shifted to the high band in proportion to the gain crossover
frequency fgc 2 ′.
[0105]
Note that, in FIG. 12, when adjustment is made to lower the gain in the feedback control filter 15,
the gain characteristic G1 becomes a characteristic indicated by the gain characteristic G3. At
this time, the gain crossover frequency fgc3 in the gain characteristic G3 is a frequency larger
than the gain crossover frequency fgc1. That is, when adjustment is made to lower the gain in the
feedback control filter 15, the gain characteristic changes from the gain characteristic G1 to the
gain characteristic G3, and the gain crossover frequency fgc1 rises to the gain crossover
frequency fgc3. Further, when the feedback control filter 15 performs adjustment to further
lower the gain, the gain characteristic G1 becomes a characteristic shown by the gain
characteristic G4. According to the gain characteristic G4, the gain is always a negative value
over the entire frequency band. Thereby, when the gain characteristic is G4, the feedback
processing is completely stabilized. However, the reduction in the feedback gain reduces the
effect of reducing distortion. The distortion reduction effect due to the gain characteristics G3
and G4 is also reduced in the speaker device 2 shown in FIG. Further, in the control system using
the speaker 16, the phase does not reach 180 °, and the phase crossover frequency fpc does
not exist. The same can be said for the speaker devices 1 to 3. Further, since the phase does not
reach 180 °, the above-mentioned phase margin always has a negative value.
[0106]
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32
As described above, according to the speaker device 3 shown in FIG. 11, by arranging the nonlinear component removal filter 10 in the feedback loop, the minimum resonance frequency f0 of
the speaker 16 is smaller than that of the speaker device 2 shown in FIG. The change in As the
variation of the lowest resonance frequency f0 of the speaker 16 becomes smaller, the variation
of the gain crossover frequency fgc also becomes smaller. Thereby, in the speaker device 3
shown in FIG. 11, even if the input is large, the distortion removal effect can be exhibited to a
frequency band lower than that of the speaker device 2 shown in FIG.
[0107]
Note that, as shown in FIG. 15, a compensation filter 21 may be further added to the front stage
of the non-linear removal filter 10 in the speaker device 3 shown in FIG. 11. FIG. 15 is a block
diagram showing a configuration example in which the compensation filter 21 is added to the
speaker device 3 shown in FIG.
[0108]
The compensation filter 21 is for increasing the low frequency level in the open loop transfer
characteristic of the speaker device 3. That is, it corresponds to the low pass filter in the present
invention. Specifically, the compensation filter 21 has a filter coefficient H represented by a
transfer function as shown in, for example, equation (18). H = k * (1 + 1 / (T * s)) (18) However, it
is assumed that T = 1 / (2 * π * fmax). Here, k is a gain, and fmax is an inflection frequency of
the frequency characteristic. The inflection frequency means a frequency when the slope of the
frequency characteristic changes. For example, let the inflection frequency be the frequency at
which the gain changes by 0 dB to 3 dB. The frequency characteristic of the transfer function
shown in equation (18) is the characteristic shown in FIG. FIG. 16 is a diagram showing gain
characteristics and phase characteristics of the compensation filter, and gain characteristics (G5
and G6) and phase characteristics (P5 and P6) of the speaker device 3. According to the gain
characteristic of the speaker device 3 shown in FIG. 16, the gain characteristic G5 of the dotted
line shown in FIG. 16 changes to the gain characteristic G6 shown by a solid line due to the filter
characteristic of the compensation filter 21. In addition, since the low band is raised in the
absence of the phase crossover frequency fpc, the gain crossover frequency fgc can be brought
close to DC. As a result, the frequency at which the above-described distortion removal effect is
obtained is lowered, so that the distortion removal effect can be further prevented from being
impaired at the time of large input, and the distortion removal effect can be exhibited to lower
08-05-2019
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frequency bands.
[0109]
The inflection frequency fmax is set to a frequency higher than at least the gain crossover
frequency fgc. Moreover, although the order of Formula (18) is first order, it is not limited to this.
As long as the gain crossover frequency fgc can be lowered, the transfer function may have a
first or higher order. When the order of Expression (18) becomes high, in the filter
characteristics of the compensation filter 21, the rising slope of the gain below the inflection
frequency becomes sharp. As a result, the gain crossover frequency fgc of the speaker device 3
can be lowered as the order of equation (18) increases, but the number of the order may be set
appropriately while considering the phase characteristics. . When the filter coefficient of the
compensation filter 21 is first order, the filter characteristic of the compensation filter 21
exhibits a characteristic of inclination at -6 dB / oct in the frequency band equal to or lower than
the inflection frequency.
[0110]
Note that a high pass filter 22 may be further added to the speaker device 3 shown in FIG. 11 as
shown in FIG. FIG. 17 is a block diagram showing a configuration example in which a high pass
filter 22 is added to the speaker device 3 shown in FIG.
[0111]
The high pass filter 22 is for preventing a signal lower than the gain crossover frequency fgc
from being input in advance. Therefore, at least the cutoff frequency needs to be equal to or
higher than the gain crossover frequency fgc. In addition, since the higher the order is, the better
the blocking characteristic is, the order may be selected by the convenience of design. Further,
when the filter coefficient of the high pass filter 22 is first order, the filter characteristic of the
high pass filter 22 exhibits a characteristic of inclination at +6 dB / oct in the frequency band
below the cutoff frequency. The high pass filter 22 may have a cutoff characteristic of +6 dB / oct
or more. In this case, the signal below the gain crossover frequency fgc is further blocked, and
the distortion reduction effect is not lost.
08-05-2019
34
[0112]
Note that, as shown in FIG. 18, a compensation filter 21 and a high pass filter 22 may be further
added to the speaker device 3 shown in FIG. FIG. 18 is a block diagram showing a configuration
example in which the compensation filter 21 and the high pass filter 22 are added to the speaker
device 3 shown in FIG.
[0113]
Here, the analysis results of the frequency characteristics of the speaker device 3 of FIG. 11, the
speaker device 3 to which only the high pass filter 22 of FIG. 17 is added, and the speaker device
3 to which the high pass filter 22 of FIG. It is shown in 19. FIG. 19 shows the analysis results
when the input is 20 W and 40 W, respectively.
[0114]
Among the second and third distortions shown in FIG. 19, it can be seen that the second and
third distortions of the speaker device 3 shown in FIG. 18 to which the high pass filter 22 and the
compensation filter 21 are added are the smallest. That is, as is also shown from this analysis
result, it can be seen that the speaker device 3 shown in FIG. 18 to which the high pass filter 22
and the compensation filter 21 are added is the device with the highest distortion removal effect.
[0115]
In the description of FIG. 12 described above, it has been described that the phase crossover
frequency fpc is not present, and the phase margin is always negative. Here, when both of the
gain margin and the phase margin described above become negative, the feedback processing
becomes unstable and oscillates. Therefore, in the case where the phase crossover frequency fpc
does not exist, and the phase margin always has a negative value, it becomes a problem how the
stability of the feedback processing becomes. On the other hand, verification is performed with
reference to the step response. In addition, in order to simplify, it analyzes by the feedback loop
of the speaker apparatus 2 shown in FIG. FIG. 20 is a diagram showing a feedback loop of the
speaker device 2 shown in FIG. The processing of the ideal filter 12 is a part of the feedback
processing, but focusing on the processing of the ideal filter 12 is processing of outputting the
08-05-2019
35
input electrical signal to the adder 14 and corresponds to feedforward processing. Also, the ideal
filter 12 models the actual speaker 16 that is a secondary vibration system. Therefore, although
the processing of the ideal filter 12 is always stable, it does not affect the stability of the abovementioned food back processing. Therefore, in evaluating the stability of the feedback
processing, the processing of the ideal filter 12 may not be considered.
[0116]
The step response results in the feedback loop shown in FIG. 20 are shown in FIGS. FIG. 21
shows the feedback loop shown in FIG. 20 when the stiffness kx, which is a nonlinear component
of the stiffness K (x) described above, is 20000, the phase margin is -0.849 °, and the gain
crossover frequency fgc is 5.4 Hz. It is a figure showing a step input and its response. FIG. 22 is a
diagram showing the step input and its response when the stiffness kx is 5000, the phase margin
is -1.7 °, and the gain crossover frequency fgc is 2.7 Hz in the feedback loop shown in FIG. FIG.
23 is a diagram showing the step input and its response when the stiffness kx is 1200, the phase
margin is −3.46 °, and the gain crossover frequency fgc is 1.3 Hz in the configuration shown in
FIG.
[0117]
Referring to the step responses shown in FIGS. 21-23, it can be seen that all step responses
converge with time. As a result, even if the phase crossover frequency fpc does not exist and the
phase is negative at the gain crossover frequency fgc, no oscillation occurs and it can be said that
the stability is high.
[0118]
In FIGS. 21 to 23, since analysis is performed by the feedback loop of the speaker device 2
shown in FIG. 10, when the stiffness kx is high, the gain crossover frequency fgc is also high. In
addition, when the gain crossover frequency fgc becomes high, the frequency of the convergent
waveform of the step response becomes high.
[0119]
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36
Fourth Embodiment A loudspeaker device 4 according to a fourth embodiment of the present
invention will be described with reference to FIG. FIG. 24 is a block diagram showing a
configuration example of the speaker device 4 according to the fourth embodiment. The speaker
device 4 according to the present embodiment is different from the above-described speaker
devices 1 to 3 according to the first to third embodiments in that a power amplifier 23 is further
provided. In FIG. 24, as an example, the speaker device 4 includes the non-linear component
removal filter 10, the linear filter 11, the ideal filter 12, the adder 13, the adder 14, the feedback
control filter 15, the speaker 16, the sensor 17, the pre-stage filter 20, and The power amplifier
23 is provided.
[0120]
In the practical application of the speaker devices according to the first to third embodiments
described above, a power amplifier for driving the speaker 16 is required. Here, among
components constituting the speaker device according to the first to third embodiments
described above, for example, there is a component such as the non-linear component removal
filter 10 that can not handle a high voltage during internal processing, As shown in FIG. 24, the
power amplifier 23 needs to be provided immediately before the speaker 16.
[0121]
In FIG. 24, the output signal of the adder 13 for removing non-linear distortion is amplified by
the power amplifier 23. For example, it is assumed that the gain of the power amplifier 23 is 10
and the input voltage of the speaker device 4 shown in FIG. 24 is 1V. In this case, the output
voltage from the power amplifier 23 is 10V. Here, when the input to the non-linear component
removal filter 10 is 1 V, the non-linear component removal filter 10 generates a signal for
removing non-linear distortion when the input to the speaker 16 is 1 V. Therefore, when the
output signal of the adder 13 is amplified to 10 V, there arises a problem that the magnitude of
non-linear distortion of the speaker 16 can not be matched.
[0122]
Therefore, it is necessary to adjust the scale of each parameter constituting the filter coefficient
of each component so that the output signal amplified by the power amplifier 23 corresponds to
08-05-2019
37
the level of non-linear distortion of the speaker 16. Hereinafter, the process of adjusting the scale
of each parameter is referred to as scaling process.
[0123]
Next, the operation principle of the speaker device 4 shown in FIG. 24 will be described. In the
following description, it is assumed that the gain of the power amplifier 23 is ten times. The
operation equation of the speaker 16 is expressed by the equation (8) as described above. (A0 +
Ax) * E (t) / Ze = (K0 + Kx) * x (t) + [r + (A0 + Ax) <2> / Ze] * dx (t) / dt + m * d <2 > x (t) / dt <2>
(8) Here, since the gain of the power amplifier 23 is ten times, each parameter is multiplied by
1/10. Thus, equation (8) is scaled down to a 1/10 model and becomes equation (19). 1/10・(A0
+ Ax) * E (t) / (1/10 · Ze) = 1/10 · (K0 + Kx) * x (t) + [1/10 · r + [1/10 (A0 + Ax)] <2> / (1/10 · Ze)]
* dx (t) / dt + 1/10 · m * d <2> x (t) / dt <2> (19) Organize equation (19) above Then, it becomes
like a formula (20). (A0 + Ax) * E (t) /0.1/Ze= (K0 + Kx) * x (t) + [r + (A0 + Ax) <2> / Ze] * dx (t) /
dt + m * d <2> x (t) / dt <2> (20) This represents an operation when a voltage of 10 V is applied
when the input voltage E is 1 V.
[0124]
Next, the non-linear component removal filter 10 generates a voltage Eff (t) that cancels out the
non-linear component as shown in the equation (21) from the result of the above equation (13).
Eff (t) = [E (t) -Ze / (A0 + Ax) * (Ax / Ze * E (t)-(2 * A0 * Ax + Ax <2>) / Ze * dx (t) / dt] −Kx * x (t))
(21) Here, considering the same as equation (19), when the input voltage E is 1 V, nonlinear
distortion corresponding to the operation of the speaker to which a voltage of 10 V is applied In
order to obtain an output for removing, each parameter of equation (21) may be multiplied by
1/10. Therefore, equation (21) becomes equation (22). Eff (t) = [E (t) − (1/10 · Ze) / [1/10 · (A0
+ Ax)] * [(1/10 · Ax) / (1/10 · Ze) * E ( t)-(2 * 1 / 10.A0 * 1 / 10.Ax + (1 / 10.Ax) <2>)] / (1 /
10.Ze) * dx (t) /dt-1/10.Kx * x (t)] ... (22) Furthermore, when the above equation (22) is arranged,
it becomes like equation (23). Eff (t) = [E (t) /0.1-Ze/ (A0 + Ax) * (Ax / Ze * E (t) /0.1- (2 * A0 * Ax
+ Ax <2>) / Ze * dx ( t) / dt−Kx * x (t)) (23) The operation of the speaker 16 to which the voltage
Eff (t) represented by the equation (23) is input is the equation (24) from the equation (13).
become that way. (A0 + Ax) / Ze * [E (t) /0.1-Ze/ (A0 + Ax) * (Ax / Ze * E (t) /0.1- (2 * A0 * Ax +
Ax <2>) / Ze * dx (t) / dt-Kx * x (t)] = (K0 + Kx) * x (t) + [r + (A0 + Ax) <2> / Ze] * dx (t) / dt + m * d
<2> x (t) / dt <2> (24) That is, assuming that the input voltage E (t) is 1 V, E (t) /0.1 is 10 V. The
same operation and processing as the operation and processing when the amplified voltage is
applied to 10 V is performed by this, and so-called scaling processing becomes possible.
08-05-2019
38
[0125]
Therefore, assuming that the gain of the power amplifier 23 is G, it may be sufficient to multiply
each parameter by 1 / G as in equation (25) when scaling processing is performed. Eff (t) = [E (t)(1 / G.Ze) / [1 / G. (A0 + Ax)] * [(1 / G.Ax) / (1 / G.Ze) * E ( t)-(2 * 1 / G.A0 * 1 / G.Ax + (1 / G.Ax)
<2>)] / (1 / G.Ze) * dx (t) /dt-1/G.Kx * x (t)) ... (25)
[0126]
The same scaling processing as the above-described non-linear removal filter 10 may be
performed for the pre-stage filter 20, the ideal filter 12, and the linear filter 11.
[0127]
As described above, when the power amplifier 23 is disposed immediately in front of the speaker
16 by performing the scaling process, the magnitude of the output voltage of the nonlinear
distortion removal filter 10 is input to the speaker 16 output from the power amplifier 23 It can
correspond to the magnitude of the voltage.
In addition, the feedforward processing unit such as the nonlinear distortion removal filter 10
can cope with the case where the voltage that can be internally processed in practice is limited.
[0128]
Furthermore, FIG. 25 is a diagram comparing frequency characteristics with and without scaling
processing. As shown in FIG. 25, it can be seen that the level of second-order and third-order
distortion becomes smaller when the scaling process is performed, and the distortion removal
effect becomes higher. This is because the addition of the power amplifier 23 to the feedback
processing unit increases the feedback gain, and the same effect as the effect described in the
gain characteristic G2 of FIG. 12 is obtained.
[0129]
08-05-2019
39
As shown in FIG. 26, the volume of the power amplifier 23, the non-linear component removal
filter 10, the linear filter 11, the ideal filter 12, the feedback control filter 15, and the pre-filter
20 are interlocked to configure volume information Vol. It may be made to reflect to the
department. Thereby, the coefficient 1 / G in the above equation (25) can be adaptively changed.
The volume information Vol indicates information on the value of the gain.
[0130]
In the speaker devices 1 to 4 described in the first to fourth embodiments, the limiter 24 may be
further provided. This can prevent the speaker 16 from being damaged due to a large input. FIG.
27 is a block diagram showing an example of a configuration in which the limiter 24 is provided
in the speaker device 1 shown in FIG. In FIG. 27, the limiter 24 limits the level of the input signal
below the level at which the speaker 16 breaks. Therefore, even if a large input signal is input,
the level higher than the level set by the limiter 24 is not input to the speaker 16, and the
speaker 16 can be prevented from being damaged. The position of the limiter 24 is not limited to
the position shown in FIG. 27. For example, the position of the limiter 24 may be between the
output of the non-linear component removing filter 10 and the input of the adder 13. It may be
between the input. That is, the limiter 24 may be disposed at any position as long as it is
disposed at a position where the input of the speaker 16 can be limited.
[0131]
In the speaker devices 1 to 4 described in the first to fourth embodiments, the non-linear
component removal filter 10, the linear filter 11, the ideal filter 12, the adder 13, the adder 14,
the feedback control filter 15, and the pre-filter 20 The compensation filter 21, the high pass
filter 22, the power amplifier 23, and the limiter 24 may be configured as an integrated circuit.
At this time, the integrated circuit includes an output terminal for outputting to the speaker 16, a
first input terminal for inputting an electrical signal, and a second input terminal for receiving a
detection signal of the sensor 17. As described above, in the first to fourth embodiments
described above, the electric circuit that performs each of the functions described above is
integrated into one small package to form, for example, an audio signal processing circuit DSP
(Digital Signal Processor) or the like. The present invention can be realized. Alternatively, the
non-linear component removal filter 10, the linear filter 11, and the ideal filter 12 may be
configured as integrated circuits, and each function may be configured as a DSP. It is effective
when the processing time of the DSP adversely affects the feedback processing and the effect is
diminished.
08-05-2019
40
[0132]
The speaker device according to the present invention can be applied to applications such as a
speaker device that can perform more stable distortion removal processing, a thin speaker, and
the like by performing signal processing that follows changes in parameters in an actual speaker.
[0133]
Block diagram showing a configuration example of the speaker device 1 according to the first
embodiment A sectional view of a general speaker 16 A diagram showing an example of
characteristics of force coefficient Bl with respect to vibration displacement x near the magnetic
gap 165 Support system for vibration displacement x A graph showing an example of a
characteristic of stiffness K A graph showing a change of a characteristic of stiffness K with
respect to an input signal I (t) A graph showing a desired output characteristic set as a filter
coefficient of the ideal filter 12 The block diagram which shows the structural example of the
speaker apparatus 1 at the time of referring the output signal of 17 The block diagram which
shows the structural example of the speaker apparatus 2 which concerns on 2nd Embodiment
The structure which changed the input of the linear filter 11 shown in FIG. Block diagram
showing an example A block diagram showing a configuration example of the speaker device 2
when the non-linear component removal filter 10 refers to the output signal of the sensor 17
Block diagram showing a configuration example of the speaker device 3 according to the third
embodiment Diagram showing gain characteristics and phase characteristics of the speaker
device 3 Diagram showing a configuration used for analysis of frequency characteristics of the
speaker device 2 shown in FIG. 13 shows a configuration example in which a compensation filter
21 is added to the speaker device 3 shown in FIG. 11 showing gain characteristics, second-order
distortion characteristics, and third-order distortion characteristics when the size of the input to
the speaker 16 is changed. 11 is a block diagram showing the frequency characteristic of the
transfer function shown in the schematic diagram (18). FIG. 11 is a block diagram showing an
example of the configuration in which a high pass filter 22 is added to the speaker device 3
shown in FIG. FIG. 1 is a block diagram showing an example of the configuration in which the
filter 21 and the high pass filter 22 are added FIG. 1 shows the analysis results when the input is
20 W and 40 W FIG. 20 shows a feedback loop of the speaker device 2 shown in FIG. 20 shows a
step input and its response in the feedback loop shown in FIG. 20 a feedback loop shown in FIG.
20 showing a step input and its response Figure showing a step input and its response block
diagram showing a configuration example of the speaker apparatus 4 according to the fourth
embodiment Figure comparing frequency characteristics with and without scaling processing
Volume of the power amplifier 23 interlocks with each component Block diagram showing an
example of a configuration in which a limiter 24 is provided to the speaker device 1 shown in
FIG. 1 showing a configuration example FIG. 1 is a block diagram showing a conventional
08-05-2019
41
speaker device 9
Explanation of sign
[0134]
DESCRIPTION OF SYMBOLS 1, 2 Speaker apparatus 10 Non-linear component removal filter 11
Linear filter 12 Ideal filter 13, 14 Adder 15 Feedback control filter 16 Speaker 17 Sensor 20 Prefilter 21 Compensation filter 22 High-pass filter 23 Power amplifier 24 Limiter 161 Voice coil
162 Diaphragm 163 Magnet 164 Magnetic circuit 165 Magnetic gap 166 Damper 167 edge
08-05-2019
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